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Strong low-pass filter

Discussion in 'Electronic Design' started by pw, Feb 14, 2013.

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  1. pw

    pw Guest

    I want to make a low-pass filter (-0.1 dB/20kHz, -150dB/22050Hz) for
    digital audio 44.1...192kHz (from S/PDIF).
    What use for this ?
    DSP processor OR FPGA OR something other?
    What is the easiest way to do this?

    Thanks in advance
    Pawel
     
  2. amdx

    amdx Guest

    Could you tell the difference between -130db and -150db?

    Mikek
     
  3. pw

    pw Guest

    If you want -130, it might be the -130.
     
  4. pw

    pw Guest

    Tim Wescott pisze:
    Please for more details. Which DSP have free CAD software available for
    the PC?
    I checked VisualDSP++ - it is not free.
    The problem is that probably all digital filters in the audio DAC's are
    full stop-band attenuation at ~0.54fs.
    In the 0.5 ... 0.54fs suppression is not full and aliasing occur.
    I want to avoid this by the use of low pass filter prior.
    If this is a bad solution - how resolve this different?
     
  5. Martin Brown

    Martin Brown Guest

    Be careful what you wish for - a brick wall filter has a horrible
    transient response. You might be better off using a notch reject filter
    to remove the 22.05kHz birdie and a much softer low pass filter.
     
  6. Hmm I think it's 20dB :^)

    George H.
     
  7. Great! I was waiting for your reply. Even with digital techniques
    you can't get around causality. (The Kramers-Kronig relations) Can
    you really make such a sharp filter... Mostly I'd just want to hit it
    with a step and see the response.

    George H.
     
  8. pw

    pw Guest

    Does the digital audio stream contain any significant content at 0.46-0.5
    I checked - the content is to 0.5fs.
    In VLC and Foobar (with PPHS resampler 44.1-->48 in ultra mode) is
    strong low-pass filter.
    I want to reproduce similar filter on the DSP or otherwise.
     
  9. Martin Brown

    Martin Brown Guest

    That last 20dB might well make the difference here between an algorithm
    that could be implemented in 32bit arithmetic realistically and
    something that can't. Sounds decidedly like audiophool stuff to me.
     
  10. mike

    mike Guest

    If you did that, what would the phase plot look like in the passband?
    Amplitude is only part of the problem with most applications???
     
  11. Jeroen

    Jeroen Guest

    That's funny. When I told some physicists here that to equalize the
    frequency response of two instruments here they just had to filter
    the signal a bit after A-to-D conversion, with a pole here and a zero
    there, that's exactly what they did. No need to mention that the
    response was so horrible that the results were useless. And it
    was _slow_ besides.

    I came up with an IIR filter that did it in three lines of code and
    on the fly, and with the right phase response too. Oh well.

    Jeroen Belleman
     
  12. Jeroen

    Jeroen Guest

    There are four parameters to play with in filter design: Complexity,
    steepness, ringing and delay. To improve on one, you'll have to
    sacrifice some or all of the others. That's true for the analog
    as well as for the digital domain, although for the latter domain,
    you can go to a degree of complexity that's undreamed of in the
    former.

    The world is full of compromise, as the OP will soon find out, no doubt.

    Jeroen Belleman
     
  13. Guest

    While the 48 --> 44.1 kHz conversion definitively needs some low pass
    filtering, why would the 44.1 -- > 48 kHz need any anti-aliasing
    filtering ?
     
  14. pw

    pw Guest

    pisze:
    I poorly know to answer to you, but I noticed that the best results in
    the resampling 44.1 -> 48 in Foobar 0.9 gives the SSRC in "ultra" mode.
    In this case the edge is the steepest. Aliasing bandwidth (22050 .. ~
    22,055) is only a few Hz. I think the slope is probably important.
    http://imageshack.us/download/560/whitenoiseconvertedfrom.png
     
  15. pw

    pw Guest

  16. pw

    pw Guest

    Vladimir Vassilevsky pisze:
    This must be realized without PC.
     
  17. amdx

    amdx Guest

    Good job, George.

    What I meant was can you measure and see the difference between -130db
    and -150 db.
    Also "(-0.1 dB/20kHz, -150dB/22050Hz)" I read that as;
    I want a filter with a passband loss of 0.1db at 20Khz and at 20.2050Khz
    I want the signal down 150db.

    Do I understand the spec?

    I don't think my neighbors TV horizontal osc. output is down 150db at my
    house 100ft away. :)
     
  18. Big grin.. of course I was just making a joke.
    Personally I'm not sure I can measure -130 or -150 dB. I've got a
    spendy SRS770 that has ~90dB of dynamic range.

    George H.
     
  19. Thanks for the heads-up on that document, Jan, it's really nice! I'm
    surprised (and a bit annoyed) that I haven't run across it before.

    For what it's worth I've collected a lot of LGPL/BSD/MIT-licensed
    filter implementations in one library here: http://www.ke5fx.com/dsplib.zip
    , with prototypes in dsplib.h. It has another Kaiser FIR generator
    implementation -- specifically my C translation of the Java code at
    dsptutor.freeuk.com, now apparently deceased.

    Anyway, check out the routines in that zipfile if you plan to go much
    farther down that path.

    -- john, KE5FX
     
  20. rickman

    rickman Guest

    The resolution of digital representations isn't all about the analog
    dynamic range. Signal processing does calculations which often require
    rounding of results. So noise is introduced which is proportional to
    the resolution of the numbers used, but not necessarily equal to that
    resolution. If you perform N successive rounding operations which are
    uncorrelated, you can expect to see N/2 lsbs of noise to be added.
    Depending on the processing being done, this can add up to the loss of
    several equivalent bits. So the idea of going beyond 24 bits is not
    obviously absurd.

    However, the proof of the pudding is in the eating. I remember years
    ago a friend showed me with a simple A/B comparison how poor cassette
    tapes were, even with Dolby noise reduction and all the other bells and
    whistles. I've yet to hear the difference between a 16 bit CD recording
    and any other representation.

    At some point you are much more limited by the rest of the system I
    think. Can we make mics and speakers with under -120 dB of distortion
    and noise? What about ears? Sometimes I can't get past the accents on
    Downton Abbey. I need the direct digital signal of closed captioning.
     
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